- Joined
- Nov 26, 2012
- Messages
- 9
- Reaction score
- 0
I am trying out 3CX with the trial (free) version. I have bought two phones, Polycom 331's and am running 3CX on a small dedicated computer running Windows 7 Pro. I have a fixed IP address from my cable provider and have purchased two SIP trunks from Nextiva. Nextiva informed me that they use port 5062 so I changed the standard 5060 to that everywhere it shows up in the system.
I have forwarded port 5090 (TCP), ports 5060-5062 (TCP+UDP) and ports 9000-9049 (UDP) to the 3cx server, which has a fixed local IP address of 192.168.1.98.
I have configured the 331 phones with fixed local IPs of 192.168.1.100 (for extension 100) and 192.168.1.102 (for extension 102). There is no extension 101 (yet).
I have set up the phones manually via their web interface.
My problem is that I can receive calls from an outside number just fine, but any attempt to call a number from either extension, whether to an ouside number or the other extension, just results in a fast busy.
I have configured a single outbound rule as follows:
Rule Name: Leading 9
Calls to Numbers starting with (Prefix) : 9
Calls from extension(s): blank
Calls to Numbers with a length of: blank
Calls from extension group: blank
Route 1: Nextiva Strip Digits: 1 Prepend: blank
Routes 2 and 3 are blank (that is, set up as block calls).
LOG RESULTS
Every so often there is the following entry in the log:
11:24:32.676 [CM506001]: STUN request to resolve SIP external IPort mapping is sent to STUN server [ V4 199.192.206.228:3478 UDP target domain=unspecified mFlowKey=0 ] over Transport [ V4 192.168.1.98:5062 UDP target domain=unspecified mFlowKey=0 ]
Very frequently I get the following log entries:
11:24:30.155 [CM302001]: Authorization system can not identify source of: SipReq: SUBSCRIBE [email protected] tid=2b3a20c41A6E4EA9 cseq=SUBSCRIBE [email protected]:5062 / 1 from(wire)
11:24:25.257 [CM302001]: Authorization system can not identify source of: SipReq: SUBSCRIBE [email protected] tid=43519c00A02D9B51 cseq=SUBSCRIBE [email protected]:5062 / 1 from(wire)
When I attempt do dial from extension 100 to extension 102 I get the following log entries:
11:26:46.097 [CM502001]: Source info: From: "100"<sip:[email protected]>;tag=A6E5E0A0-D0B11D35<sip:[email protected];user=phone>
11:26:46.097 [CM503013]: Call(30): Incoming call rejected, caller is unknown; msg=SipReq: INVITE [email protected] tid=bbefbdf72337E5B8 cseq=INVITE [email protected]:5062 / 2 from(wire)
11:26:45.679 [CM500002]: Unidentified incoming call. Review INVITE and adjust source identification:
INVITE sip:[email protected];user=phone;transport=tcp SIP/2.0
Via: SIP/2.0/TCP 192.168.1.100:5062;branch=z9hG4bKd492308e590FB47B
Max-Forwards: 70
Contact: <sip:[email protected]:5062;transport=tcp>
To: <sip:[email protected];user=phone>
From: "100"<sip:[email protected]>;tag=A6E5E0A0-D0B11D35
Call-ID: [email protected]
CSeq: 1 INVITE
Accept-Language: en
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Supported: 100rel, replaces
User-Agent: PolycomSoundPointIP-SPIP_331-UA/4.0.3.7562
Allow-Events: conference, talk, hold
Content-Length: 0
When I attempt to dial an outside number such as my cell phone I get the following log entries (last four digits of the number have been changed to 'nnnn'):
11:28:39.324 [CM502001]: Source info: From: "100"<sip:[email protected]>;tag=D1FFFFC5-8ED3C05E<sip:[email protected];user=phone>
11:28:39.324 [CM503013]: Call(31): Incoming call rejected, caller is unknown; msg=SipReq: INVITE [email protected] tid=395847f2BCC5962F cseq=INVITE [email protected]:5062 / 2 from(wire)
11:28:38.892 [CM500002]: Unidentified incoming call. Review INVITE and adjust source identification:
INVITE sip:[email protected];user=phone;transport=tcp SIP/2.0
Via: SIP/2.0/TCP 192.168.1.100:5062;branch=z9hG4bK734f0ead9CE0BA66
Max-Forwards: 70
Contact: <sip:[email protected]:5062;transport=tcp>
To: <sip:[email protected];user=phone>
From: "100"<sip:[email protected]>;tag=D1FFFFC5-8ED3C05E
Call-ID: [email protected]
CSeq: 1 INVITE
Accept-Language: en
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Supported: 100rel, replaces
User-Agent: PolycomSoundPointIP-SPIP_331-UA/4.0.3.7562
Allow-Events: conference, talk, hold
Content-Length: 0
I have searched the forum entries as best I could before posting, but I either found postings that had the same or a similar symptom but a different source problem, or else the suggested solution was one I'd already tried or implemented at first.
Any help would be greatly appreciated. I've spend quite a few hours on this and am a bit tired of hearing that fast busy!
I have forwarded port 5090 (TCP), ports 5060-5062 (TCP+UDP) and ports 9000-9049 (UDP) to the 3cx server, which has a fixed local IP address of 192.168.1.98.
I have configured the 331 phones with fixed local IPs of 192.168.1.100 (for extension 100) and 192.168.1.102 (for extension 102). There is no extension 101 (yet).
I have set up the phones manually via their web interface.
My problem is that I can receive calls from an outside number just fine, but any attempt to call a number from either extension, whether to an ouside number or the other extension, just results in a fast busy.
I have configured a single outbound rule as follows:
Rule Name: Leading 9
Calls to Numbers starting with (Prefix) : 9
Calls from extension(s): blank
Calls to Numbers with a length of: blank
Calls from extension group: blank
Route 1: Nextiva Strip Digits: 1 Prepend: blank
Routes 2 and 3 are blank (that is, set up as block calls).
LOG RESULTS
Every so often there is the following entry in the log:
11:24:32.676 [CM506001]: STUN request to resolve SIP external IPort mapping is sent to STUN server [ V4 199.192.206.228:3478 UDP target domain=unspecified mFlowKey=0 ] over Transport [ V4 192.168.1.98:5062 UDP target domain=unspecified mFlowKey=0 ]
Very frequently I get the following log entries:
11:24:30.155 [CM302001]: Authorization system can not identify source of: SipReq: SUBSCRIBE [email protected] tid=2b3a20c41A6E4EA9 cseq=SUBSCRIBE [email protected]:5062 / 1 from(wire)
11:24:25.257 [CM302001]: Authorization system can not identify source of: SipReq: SUBSCRIBE [email protected] tid=43519c00A02D9B51 cseq=SUBSCRIBE [email protected]:5062 / 1 from(wire)
When I attempt do dial from extension 100 to extension 102 I get the following log entries:
11:26:46.097 [CM502001]: Source info: From: "100"<sip:[email protected]>;tag=A6E5E0A0-D0B11D35<sip:[email protected];user=phone>
11:26:46.097 [CM503013]: Call(30): Incoming call rejected, caller is unknown; msg=SipReq: INVITE [email protected] tid=bbefbdf72337E5B8 cseq=INVITE [email protected]:5062 / 2 from(wire)
11:26:45.679 [CM500002]: Unidentified incoming call. Review INVITE and adjust source identification:
INVITE sip:[email protected];user=phone;transport=tcp SIP/2.0
Via: SIP/2.0/TCP 192.168.1.100:5062;branch=z9hG4bKd492308e590FB47B
Max-Forwards: 70
Contact: <sip:[email protected]:5062;transport=tcp>
To: <sip:[email protected];user=phone>
From: "100"<sip:[email protected]>;tag=A6E5E0A0-D0B11D35
Call-ID: [email protected]
CSeq: 1 INVITE
Accept-Language: en
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Supported: 100rel, replaces
User-Agent: PolycomSoundPointIP-SPIP_331-UA/4.0.3.7562
Allow-Events: conference, talk, hold
Content-Length: 0
When I attempt to dial an outside number such as my cell phone I get the following log entries (last four digits of the number have been changed to 'nnnn'):
11:28:39.324 [CM502001]: Source info: From: "100"<sip:[email protected]>;tag=D1FFFFC5-8ED3C05E<sip:[email protected];user=phone>
11:28:39.324 [CM503013]: Call(31): Incoming call rejected, caller is unknown; msg=SipReq: INVITE [email protected] tid=395847f2BCC5962F cseq=INVITE [email protected]:5062 / 2 from(wire)
11:28:38.892 [CM500002]: Unidentified incoming call. Review INVITE and adjust source identification:
INVITE sip:[email protected];user=phone;transport=tcp SIP/2.0
Via: SIP/2.0/TCP 192.168.1.100:5062;branch=z9hG4bK734f0ead9CE0BA66
Max-Forwards: 70
Contact: <sip:[email protected]:5062;transport=tcp>
To: <sip:[email protected];user=phone>
From: "100"<sip:[email protected]>;tag=D1FFFFC5-8ED3C05E
Call-ID: [email protected]
CSeq: 1 INVITE
Accept-Language: en
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Supported: 100rel, replaces
User-Agent: PolycomSoundPointIP-SPIP_331-UA/4.0.3.7562
Allow-Events: conference, talk, hold
Content-Length: 0
I have searched the forum entries as best I could before posting, but I either found postings that had the same or a similar symptom but a different source problem, or else the suggested solution was one I'd already tried or implemented at first.
Any help would be greatly appreciated. I've spend quite a few hours on this and am a bit tired of hearing that fast busy!